1. Field
Methods and apparatuses consistent with exemplary embodiments relate to a signal processing apparatus, which receives a mobile voice over IP (VoIP)-based voice packet and processes the voice packet, and a signal processing method thereof.
2. Description of the Related Art
If voice data is transmitted through a VoIP-based packet network, packets arrive at a receiving end at irregular intervals, because each packet is transmitted through a different path rather than a same path. In other words, the packets may be lost due to a network load, and the packets may not be received within a fixed time, or may be received in a different order from that when they are transmitted. Therefore, sound quality may deteriorate.
In order to estimate jitter, which may cause such sound quality deterioration, a related-art method transmits a test packet to a network path and estimates jitter using a time stamp and sequence information of the test packet. Another related-art method for estimating network jitter calculates an average and a variance of jitter, using information on packets arriving at a receiving end, multiplies the variance by a fixed jitter variance weighting, and adds the product of the variance and the weighing and the average of the jitter, thereby estimating network jitter.
A related-art playout method to prevent voice quality from deteriorating due to a delay and jitter generated in a network, measures a network traffic during a predetermined time period, compares the network traffic with a threshold value, and performs one signal processing operation from among expanding, compression, and normal output with respect to a packet existing in a receiving end, thereby reducing a buffering delay. If a jitter buffer is expected to be used up, since a network delay is abruptly increased or the jitter buffer is expected to receive packets out of its packet receiving range, i.e., is expected to suffer from a underflow or overflow phenomenon since the network delay is abruptly reduced, the playout method performs expanding and compression.
Also, a related-art packet loss concealing method, which conceals a packet that is lost in a network and does not arrive at a receiving end, estimates a voice parameter regarding the lost packet based on a normally decoded previous packet, and decodes and uses the lost voice packet using the estimated parameter, or restores a lost voice frame by increasing pitch of a previous voice frame one by one using information of the previous frame of the lost frame.
However, the above-described methods may increase the buffering delay, and may cause sound quality to seriously deteriorate in a network situation which is dynamically changed or an abnormal network situation, such as spike. Also, during the signal processing operations such as compression, expanding, loss concealment, and combination, the sound quality may be damaged and may deteriorate. Accordingly, there is a demand for a method for solving an increased buffering delay and sound quality deterioration.